I followed these instructions for Mod_vmd except for a Windows box:
http://wiki.freeswitch.org/wiki/Mod_vmd
I tried testing to see if it's working by dialing the following extension:
??? !-- mod_vmd test extension (new mod)--
??? extension name=vmdtest
??? ??? condition field=destination_number
Mark,
It does work ... but I can't really attest to how well ... especially
compared to other things out there. I started capturing this in CDR's to
see and it didn't seem like it worked very well.
If this is really critical to you, you might want to ping Ken Rice. I know
he might have a
Hello
For single-host settings, getting customers to buy a separate server
just to run Freeswitch is overkill, so I'm thinking about selling
just the IVR application to run on Windows. Unless a PCI card is
available, the FXO connection will be provided by Sangoma's USB device.
I'd like some
thanks Mathieu.
I setup an IRC account to give it a try.
Comme ça je pourrais t'embeter avec mes pbms :p
rod
Mathieu Rene wrote:
limit_hash uses a faster data structure then limit but works the same
way for tne end-user.
viens sur IRC si t'as des questions en francais =)
Math
On
To share my experience: I had issues with echo with many E1 trunks in
Serbia, especially when voice in between telco's network went to well known
bad analog lines. I used OSLEC and I was fortunate to have Steve to complain
to, he helped patching it further after my beta testing. Not many people
Hello,
ok found it ... was a configuration issue due to the continue on fail =
true variable in my dialplan. Hangup application fixed this :)
Sorry for the post.
regards
helmut
On 18.03.2009 10:20, Helmut Kuper wrote:
Hello,
I'm not sure whether the following is a bug or a config issue:
Hello,
I'm not sure whether the following is a bug or a config issue:
I found this in my log file:
2009-03-18 10:07:00 [INFO] mod_dptools.c:1998 audio_bridge_function()
Originate Failed. Cause: USER_BUSY
2009-03-18 10:07:00 [DEBUG] mod_dptools.c:2025 audio_bridge_function()
Continue on fail
Hi,
I've recently ugrade to version 1.02 of freeswitch and am having some
problems with my gateway registrations. The gateway successfully registers
with my voip provider when freeswitch first starts but if left running it
seems to loose it's connection to my voip provider. I can get it to
SDR?
I'm wondering why there was nothing in the console showing the channel
variable ${vmd_detect} as the wiki says there should be:
action application=info/ !-- Look for chan var vmd_detect here --
Mark
-Original Message-
From: Shelby Ramsey sicfsl...@gmail.com
To:
Mark,
Because it didn't detect a beep. It will be be there as vmd_detect=true
if it does. I'm not sure exactly how reliable it's beep detection is.
SDR
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OK, one last go and I give up.
Lets look at the documentation for Dialogic springware. This is the DSP
package that loads in their cards or runs on the host in HMP
applications. It does things like DTMF generation and detection for all
Dialogic cards except the DM3 series. The documentation
Another issue with this module is the resources it consumes. We had it
running on 50 calls yesterday and the cpu's all went to 90+%
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On
Nik Middleton wrote:
Another issue with this module is the resources it consumes. We had it
running on 50 calls yesterday and the cpu’s all went to 90+%
That's odd. Something must be fouling up, as the algorithm he used
should be very lightweight.
Steve
Upgrade to 1.03 or SVN Trunk
/b
On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote:
Hi,
I've recently ugrade to version 1.02 of freeswitch and am having
some problems with my gateway registrations. The gateway
successfully registers with my voip provider when freeswitch first
starts but if
if you are behind NAT it is possible that your router forgot the
mapping betweeen FS and your provider, try addingparam
name=ping value=30 / to your gateway.
Math
On 18-Mar-09, at 10:07 AM, Brian West wrote:
Upgrade to 1.03 or SVN Trunk
/b
On Mar 18, 2009, at 6:20 AM, Andy Ayers
On Mar 17, 2009, at 10:31 PM, Jason White wrote:
Brian West br...@freeswitch.org wrote:
if you installed the ssl devel stuff AFTER you configured you'll need
to reconfigure.
I'm reasonably sure it was installed already, unless it was pulled
in recently
by a package upgrade.
The
There is currently no openzap (sangoma, etc) support on windows, we
hope this will be coming soon.
Mike
On Mar 17, 2009, at 5:20 AM, Gilles wrote:
Hello
For single-host settings, getting customers to buy a separate server
just to run Freeswitch is overkill, so I'm thinking about selling
I added a voicemail tag in to a default extension 1001, I hear the
voicemail beep but still don't see vmd_detect.
Mark
-Original Message-
From: Shelby Ramsey sicfsl...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 6:07 am
Subject: Re:
We're a couple more steps forward from yesterday. Turned out some of my regex
was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra
space before one of the closing brackets in the default.xml example. After
staring at the screen all day it's funny how you miss these things!
We're a couple more steps forward from yesterday. Turned out some of my
regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has
an extra space before one of the closing brackets in the default.xml
example. After staring at the screen all day it's funny how you miss
these things!
I upgraded to
FreeSWITCH Version 1.0.trunk (12654M)
but caller is still being hungup (and not continuing on with dialplan) after
agent disconnect with hangup_after_bridge=false
Is there a separate patch I need to apply? Thanks.
--matt
On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong
2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch
there was a timer problem which was not solved yet. This caused channels
to be busy in my case.
I am not sure whether this is solved yet. Can anybody confirm?
Best regards
Peter
Mark Tabron schrieb:
We're a couple more
2009/3/18 mszla...@aol.com:
I added a voicemail tag in to a default extension 1001, I hear the
voicemail beep but still don't see vmd_detect.
Mark
FYI, I've used mod_vmd but only in a TDM environment on outbound calls
via a PRI. It worked very well on for detecting answering machine
On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX prometheus...@gmx.net wrote:
2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch
there was a timer problem which was not solved yet. This caused channels
to be busy in my case.
I am not sure whether this is solved yet. Can
This is the patch
http://jira.freeswitch.org/browse/MODAPP-237
it's not added yet.
2009/3/18 Matthew Fong mattdf...@gmail.com
I upgraded to
FreeSWITCH Version 1.0.trunk (12654M)
but caller is still being hungup (and not continuing on with dialplan)
after agent disconnect with
Hi MC,
With trunk 12638M, I tried checking vmd internally and externally to my cell.
No luck at all in detecting a voicemail (beep).
I used the following extensions to test this, maybe they are in error.
If not then how else can I detect from FS that I got voicemail in a phone
agnostic way
Was this ever resolved?
If we're missing something in the documentation, I'd like to make sure
it's in there.
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3239 x0
On Mar 18, 2009, at 11:34 AM, Michael Jerris wrote:
On Mar 17, 2009, at 10:31 PM, Jason White wrote:
Brian
I thought we had... hrm.
/b
On Mar 18, 2009, at 5:39 PM, Karl Vesterling wrote:
Was this ever resolved?
If we're missing something in the documentation, I'd like to make
sure it's in there.
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3239 x0
Hmm,
Well We're connected direct to E1's and it doesn't work reliably here.
That said, DTMF detect does recognise the beeps most of the time.
Perhaps there's a regional variation. I wonder if it's country
specific. The code looks logical. When I get some time I'll have a
look at it and see how
Ironically, I've used tone_detect to try and trap SIT tones and I
found that answering machines in the USA seem to all send a beep in
the same freq range as American SIT tones... :)
-MC
On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hmm,
Well We're
Hi all,
mod_dingaling in client mode works well for me, but disconnected
yesterday.
2009-03-18 16:57:32 [DEBUG] libdingaling.c:1545 xmpp_connect() io
error 2 7
I use dl_login profile=gmail.com, and it re-login successfully. Is
their a way to auto re-login after fail?
Thanks.
tone_detect! sounds good.
BTW, was there any errors in those extensions I posted. I modified something
you posted MC.
-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 5:15 pm
Subject: Re:
Is this issue still open? I just noticed this.
The error you are receiving indicates UnixODBC is installed, but not
configured properly (most likely anyway). The UnixODBC drivers are kind of a
pain to setup on some systems, especially CentOS, but this article may help
you get it working -
I do have auto-login enabled in jingle_profile:
param name=auto-login value=true/
param name=auto-reply value=Press *Call* to join my conference/
param name=sasl value=plain/
param name=tls value=true/
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Hi list,
Any experience building FS in Solaris using Sun Studio?
Thanks
Pablo
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